mirror of https://github.com/mpv-player/mpv
352 lines
9.2 KiB
C
352 lines
9.2 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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*/
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/* This audio filter changes the sample rate. */
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <inttypes.h>
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#include "af.h"
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#include "dsp.h"
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/* Below definition selects the length of each poly phase component.
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Valid definitions are L8 and L16, where the number denotes the
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length of the filter. This definition affects the computational
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complexity (see play()), the performance (see filter.h) and the
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memory usage. The filterlenght is choosen to 8 if the machine is
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slow and to 16 if the machine is fast and has MMX.
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*/
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#if !defined(HAVE_SSE) && !defined(HAVE_3DNOW) // This machine is slow
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#define L 8 // Filter length
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// Unrolled loop to speed up execution
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#define FIR(x,w,y) \
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(y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
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+ w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) >> 16
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#else /* Fast machine */
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#define L 16
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// Unrolled loop to speed up execution
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#define FIR(x,w,y) \
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y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
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+ w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
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+ w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
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+ w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) >> 16
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#endif /* Fast machine */
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// Macro to add data to circular que
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#define ADDQUE(xi,xq,in)\
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xq[xi]=xq[xi+L]=(*in);\
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xi=(xi-1)&(L-1);
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// local data
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typedef struct af_resample_s
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{
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int16_t* w; // Current filter weights
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int16_t** xq; // Circular buffers
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uint32_t xi; // Index for circular buffers
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uint32_t wi; // Index for w
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uint32_t i; // Number of new samples to put in x queue
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uint32_t dn; // Down sampling factor
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uint32_t up; // Up sampling factor
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int sloppy; // Enable sloppy resampling to reduce memory usage
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int fast; // Enable linear interpolation instead of filtering
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} af_resample_t;
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// Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
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static inline int gcd(register int a, register int b)
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{
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register int r = min(a,b);
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a=max(a,b);
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b=r;
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r=a%b;
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while(r!=0){
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a=b;
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b=r;
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r=a%b;
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}
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return b;
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}
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static int upsample(af_data_t* c,af_data_t* l, af_resample_t* s)
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{
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uint32_t ci = l->nch; // Index for channels
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uint32_t len = 0; // Number of input samples
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uint32_t nch = l->nch; // Number of channels
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uint32_t inc = s->up/s->dn;
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uint32_t level = s->up%s->dn;
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uint32_t up = s->up;
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uint32_t dn = s->dn;
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register int16_t* w = s->w;
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register uint32_t wi = 0;
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register uint32_t xi = 0;
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// Index current channel
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while(ci--){
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// Temporary pointers
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register int16_t* x = s->xq[ci];
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register int16_t* in = ((int16_t*)c->audio)+ci;
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register int16_t* out = ((int16_t*)l->audio)+ci;
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int16_t* end = in+c->len/2; // Block loop end
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wi = s->wi; xi = s->xi;
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while(in < end){
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register uint32_t i = inc;
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if(wi<level) i++;
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ADDQUE(xi,x,in);
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in+=nch;
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while(i--){
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// Run the FIR filter
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FIR((&x[xi]),(&w[wi*L]),out);
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len++; out+=nch;
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// Update wi to point at the correct polyphase component
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wi=(wi+dn)%up;
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}
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}
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}
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// Save values that needs to be kept for next time
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s->wi = wi;
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s->xi = xi;
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return len;
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}
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static int downsample(af_data_t* c,af_data_t* l, af_resample_t* s)
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{
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uint32_t ci = l->nch; // Index for channels
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uint32_t len = 0; // Number of output samples
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uint32_t nch = l->nch; // Number of channels
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uint32_t inc = s->dn/s->up;
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uint32_t level = s->dn%s->up;
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uint32_t up = s->up;
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uint32_t dn = s->dn;
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register int32_t i = 0;
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register uint32_t wi = 0;
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register uint32_t xi = 0;
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// Index current channel
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while(ci--){
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// Temporary pointers
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register int16_t* x = s->xq[ci];
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register int16_t* in = ((int16_t*)c->audio)+ci;
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register int16_t* out = ((int16_t*)l->audio)+ci;
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register int16_t* end = in+c->len/2; // Block loop end
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i = s->i; wi = s->wi; xi = s->xi;
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while(in < end){
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ADDQUE(xi,x,in);
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in+=nch;
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if((--i)<=0){
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// Run the FIR filter
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FIR((&x[xi]),(&s->w[wi*L]),out);
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len++; out+=nch;
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// Update wi to point at the correct polyphase component
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wi=(wi+dn)%up;
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// Insert i number of new samples in queue
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i = inc;
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if(wi<level) i++;
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}
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}
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}
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// Save values that needs to be kept for next time
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s->wi = wi;
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s->xi = xi;
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s->i = i;
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return len;
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}
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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switch(cmd){
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case AF_CONTROL_REINIT:{
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af_resample_t* s = (af_resample_t*)af->setup;
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af_data_t* n = (af_data_t*)arg; // New configureation
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int i,d = 0;
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int rv = AF_OK;
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// Make sure this filter isn't redundant
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if(af->data->rate == n->rate)
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return AF_DETACH;
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// Create space for circular bufers (if nesessary)
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if(af->data->nch != n->nch){
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// First free the old ones
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if(s->xq){
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for(i=1;i<af->data->nch;i++)
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if(s->xq[i])
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free(s->xq[i]);
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free(s->xq);
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}
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// ... then create new
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s->xq = malloc(n->nch*sizeof(int16_t*));
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for(i=0;i<n->nch;i++)
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s->xq[i] = malloc(2*L*sizeof(int16_t));
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s->xi = 0;
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}
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// Set parameters
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af->data->nch = n->nch;
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af->data->format = AF_FORMAT_NE | AF_FORMAT_SI;
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af->data->bps = 2;
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if(af->data->format != n->format || af->data->bps != n->bps)
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rv = AF_FALSE;
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n->format = AF_FORMAT_NE | AF_FORMAT_SI;
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n->bps = 2;
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// Calculate up and down sampling factors
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d=gcd(af->data->rate,n->rate);
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// If sloppy resampling is enabled limit the upsampling factor
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if(s->sloppy && (af->data->rate/d > 5000)){
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int up=af->data->rate/2;
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int dn=n->rate/2;
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int m=2;
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while(af->data->rate/(d*m) > 5000){
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d=gcd(up,dn);
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up/=2; dn/=2; m*=2;
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}
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d*=m;
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}
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// Check if the the design needs to be redone
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if(s->up != af->data->rate/d || s->dn != n->rate/d){
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float* w;
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float* wt;
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float fc;
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int j;
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s->up = af->data->rate/d;
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s->dn = n->rate/d;
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// Calculate cuttof frequency for filter
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fc = 1/(float)(max(s->up,s->dn));
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// Allocate space for polyphase filter bank and protptype filter
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w = malloc(sizeof(float) * s->up *L);
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if(NULL != s->w)
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free(s->w);
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s->w = malloc(L*s->up*sizeof(int16_t));
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// Design prototype filter type using Kaiser window with beta = 10
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if(NULL == w || NULL == s->w ||
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-1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
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af_msg(AF_MSG_ERROR,"[resample] Unable to design prototype filter.\n");
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return AF_ERROR;
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}
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// Copy data from prototype to polyphase filter
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wt=w;
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for(j=0;j<L;j++){//Columns
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for(i=0;i<s->up;i++){//Rows
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float t=(float)s->up*32767.0*(*wt);
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s->w[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
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wt++;
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}
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}
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free(w);
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af_msg(AF_MSG_VERBOSE,"[resample] New filter designed up: %i down: %i\n", s->up, s->dn);
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}
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// Set multiplier and delay
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af->delay = (double)(1000*L/2)/((double)n->rate);
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af->mul.n = s->up;
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af->mul.d = s->dn;
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return rv;
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}
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case AF_CONTROL_COMMAND_LINE:{
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af_resample_t* s = (af_resample_t*)af->setup;
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int rate=0;
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sscanf((char*)arg,"%i:%i:%i",&rate,&(s->sloppy), &(s->fast));
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return af->control(af,AF_CONTROL_RESAMPLE,&rate);
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}
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case AF_CONTROL_RESAMPLE:
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// Reinit must be called after this function has been called
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// Sanity check
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if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
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af_msg(AF_MSG_ERROR,"[resample] The output sample frequency must be between 8kHz and 192kHz. Current value is %i \n",((int*)arg)[0]);
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return AF_ERROR;
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}
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af->data->rate=((int*)arg)[0];
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af_msg(AF_MSG_VERBOSE,"[resample] Changing sample rate to %iHz\n",af->data->rate);
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data);
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}
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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int len = 0; // Length of output data
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af_data_t* c = data; // Current working data
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af_data_t* l = af->data; // Local data
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af_resample_t* s = (af_resample_t*)af->setup;
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if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
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return NULL;
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// Run resampling
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if(s->up>s->dn)
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len = upsample(c,l,s);
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else
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len = downsample(c,l,s);
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// Set output data
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c->audio = l->audio;
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c->len = len*2;
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c->rate = l->rate;
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return c;
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}
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// Allocate memory and set function pointers
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static int open(af_instance_t* af){
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul.n=1;
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af->mul.d=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=calloc(1,sizeof(af_resample_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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return AF_OK;
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}
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// Description of this plugin
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af_info_t af_info_resample = {
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"Sample frequency conversion",
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"resample",
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"Anders",
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"",
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AF_FLAGS_REENTRANT,
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open
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};
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